/*
 * Audio.h
 *
 *  Created on: Oct 28,2018
 *
 *  Version 3.0.7
 *  Updated on: Oct 08.2023
 *      Author: Wolle (schreibfaul1)
 */

#pragma once
#pragma GCC optimize("Ofast")
#include <vector>
#include <Arduino.h>
#include <libb64/cencode.h>
#include <esp32-hal-log.h>
#include <SPI.h>
#include <WiFi.h>
#include <WiFiClientSecure.h>
#include <vector>
#include <driver/i2s.h>
#include <SD.h>
#include <SD_MMC.h>
#include <SPIFFS.h>
#include <FS.h>
#include <FFat.h>

using namespace std;

extern __attribute__((weak)) void audio_info(const char *);
extern __attribute__((weak)) void audio_id3data(const char *);                                      // ID3 metadata
extern __attribute__((weak)) void audio_id3image(File &file, const size_t pos, const size_t size);  // ID3 metadata image
extern __attribute__((weak)) void audio_id3lyrics(File &file, const size_t pos, const size_t size); // ID3 metadata lyrics
extern __attribute__((weak)) void audio_eof_mp3(const char *);                                      // end of mp3 file
extern __attribute__((weak)) void audio_showstreamtitle(const char *);
extern __attribute__((weak)) void audio_showstation(const char *);
extern __attribute__((weak)) void audio_bitrate(const char *);
extern __attribute__((weak)) void audio_commercial(const char *);
extern __attribute__((weak)) void audio_icyurl(const char *);
extern __attribute__((weak)) void audio_icydescription(const char *);
extern __attribute__((weak)) void audio_lasthost(const char *);
extern __attribute__((weak)) void audio_eof_speech(const char *);
extern __attribute__((weak)) void audio_eof_stream(const char *);                                       // The webstream comes to an end
extern __attribute__((weak)) void audio_process_extern(int16_t *buff, uint16_t len, bool *continueI2S); // record audiodata or send via BT
extern __attribute__((weak)) void audio_process_i2s(uint32_t *sample, bool *continueI2S);               // record audiodata or send via BT
extern __attribute__((weak)) void audio_log(uint8_t logLevel, const char* msg, const char* arg);        //新加的

//----------------------------------------------------------------------------------------------------------------------

class AudioBuffer
{
    // AudioBuffer will be allocated in PSRAM, If PSRAM not available or has not enough space AudioBuffer will be
    // allocated in FlashRAM with reduced size
    //
    //  m_buffer            m_readPtr                 m_writePtr                 m_endPtr
    //   |                       |<------dataLength------->|<------ writeSpace ----->|
    //   ??                       ??                         ??                         ??
    //   ---------------------------------------------------------------------------------------------------------------
    //   |                     <--m_buffSize-->                                      |      <--m_resBuffSize -->     |
    //   ---------------------------------------------------------------------------------------------------------------
    //   |<-----freeSpace------->|                         |<------freeSpace-------->|
    //
    //
    //
    //   if the space between m_readPtr and buffend < m_resBuffSize copy data from the beginning to resBuff
    //   so that the mp3/aac/flac frame is always completed
    //
    //  m_buffer                      m_writePtr                 m_readPtr        m_endPtr
    //   |                                 |<-------writeSpace------>|<--dataLength-->|
    //   ??                                 ??                         ??                ??
    //   ---------------------------------------------------------------------------------------------------------------
    //   |                        <--m_buffSize-->                                    |      <--m_resBuffSize -->     |
    //   ---------------------------------------------------------------------------------------------------------------
    //   |<---  ------dataLength--  ------>|<-------freeSpace------->|
    //
    //

public:
    AudioBuffer(size_t maxBlockSize = 0); // constructor
    ~AudioBuffer();                       // frees the buffer
    size_t init();                        // set default values
    bool isInitialized() { return m_f_init; };
    void setBufsize(int ram, int psram);
    void changeMaxBlockSize(uint16_t mbs); // is default 1600 for mp3 and aac, set 16384 for FLAC
    uint16_t getMaxBlockSize();            // returns maxBlockSize
    size_t freeSpace();                    // number of free bytes to overwrite
    size_t writeSpace();                   // space fom writepointer to bufferend
    size_t bufferFilled();                 // returns the number of filled bytes
    void bytesWritten(size_t bw);          // update writepointer
    void bytesWasRead(size_t br);          // update readpointer
    uint8_t *getWritePtr();                // returns the current writepointer
    uint8_t *getReadPtr();                 // returns the current readpointer
    uint32_t getWritePos();                // write position relative to the beginning
    uint32_t getReadPos();                 // read position relative to the beginning
    void resetBuffer();                    // restore defaults
    bool havePSRAM() { return m_f_psram; };

protected:
    size_t m_buffSizePSRAM = UINT16_MAX * 10; // most webstreams limit the advance to 100...300Kbytes
    size_t m_buffSizeRAM = 1600 * 10;
    size_t m_buffSize = 0;
    size_t m_freeSpace = 0;
    size_t m_writeSpace = 0;
    size_t m_dataLength = 0;
    size_t m_resBuffSizeRAM = 1600;       // reserved buffspace, >= one mp3  frame
    size_t m_resBuffSizePSRAM = 4096 * 4; // reserved buffspace, >= one flac frame
    size_t m_maxBlockSize = 1600;
    uint8_t *m_buffer = NULL;
    uint8_t *m_writePtr = NULL;
    uint8_t *m_readPtr = NULL;
    uint8_t *m_endPtr = NULL;
    bool m_f_start = true;
    bool m_f_init = false;
    bool m_f_psram = false; // PSRAM is available (and used...)
};
//----------------------------------------------------------------------------------------------------------------------

class Audio2 : private AudioBuffer
{

    AudioBuffer InBuff; // instance of input buffer

public:
    Audio2(bool internalDAC = false, uint8_t channelEnabled = 3, uint8_t i2sPort = I2S_NUM_0); // #99
    ~Audio2();
    void setBufsize(int rambuf_sz, int psrambuf_sz);
    bool connecttohost(const char *host, const char *user = "", const char *pwd = "");
    bool connecttospeech(const char *speech, const char *lang);
    bool connecttoFS(fs::FS &fs, const char *path, int32_t resumeFilePos = -1);
    bool connecttoSD(const char *path, int32_t resumeFilePos = -1);
    bool setFileLoop(bool input); // TEST loop
    void setConnectionTimeout(uint16_t timeout_ms, uint16_t timeout_ms_ssl);
    bool setAudioPlayPosition(uint16_t sec);
    bool setFilePos(uint32_t pos);
    bool audioFileSeek(const float speed);
    bool setTimeOffset(int sec);
    bool setPinout(uint8_t BCLK, uint8_t LRC, uint8_t DOUT, int8_t DIN = I2S_PIN_NO_CHANGE, int8_t MCK = I2S_PIN_NO_CHANGE);
    bool pauseResume();
    bool isRunning() { return m_f_running; }
    void loop();
    uint32_t stopSong();
    void forceMono(bool m);
    void setBalance(int8_t bal = 0);
    void setVolumeSteps(uint8_t steps);
    void setVolume(uint8_t vol, uint8_t curve = 0);
    uint8_t getVolume();
    uint8_t maxVolume();
    uint8_t getI2sPort();

    uint32_t getAudioDataStartPos();
    uint32_t getFileSize();
    uint32_t getFilePos();
    uint32_t getSampleRate();
    uint8_t getBitsPerSample();
    uint8_t getChannels();
    uint32_t getBitRate(bool avg = false);
    uint32_t getAudioFileDuration();
    uint32_t getAudioCurrentTime();
    uint32_t getTotalPlayingTime();
    uint16_t getVUlevel();

    esp_err_t i2s_mclk_pin_select(const uint8_t pin);
    uint32_t inBufferFilled(); // returns the number of stored bytes in the inputbuffer
    uint32_t inBufferFree();   // returns the number of free bytes in the inputbuffer
    void setTone(int8_t gainLowPass, int8_t gainBandPass, int8_t gainHighPass);
    void setI2SCommFMT_LSB(bool commFMT);
    int getCodec() { return m_codec; }
    const char *getCodecname() { return codecname[m_codec]; }
    void unicode2utf8(char *buff, uint32_t len);
    uint8_t isplaying = 0;

private:
#ifndef ESP_ARDUINO_VERSION_VAL
#define ESP_ARDUINO_VERSION_MAJOR 0
#define ESP_ARDUINO_VERSION_MINOR 0
#define ESP_ARDUINO_VERSION_PATCH 0
#endif
    // 新加的
    enum : int8_t { AUDIOLOG_PATH_IS_NULL = -1, AUDIOLOG_FILE_NOT_FOUND = -2, AUDIOLOG_OUT_OF_MEMORY = -3, AUDIOLOG_FILE_READ_ERR = -4, AUDIOLOG_ERR_UNKNOWN = -127 };

    void UTF8toASCII(char *str);
    bool latinToUTF8(char *buff, size_t bufflen);
    void setDefaults(); // free buffers and set defaults
    void initInBuff();
    bool httpPrint(const char *host);
    void processLocalFile();
    void processWebStream();
    void processWebFile();
    void processWebStreamTS();
    void processWebStreamHLS();
    void playAudioData();
    bool readPlayListData();
    const char *parsePlaylist_M3U();
    const char *parsePlaylist_PLS();
    const char *parsePlaylist_ASX();
    const char *parsePlaylist_M3U8();
    bool STfromEXTINF(char *str);
    void showCodecParams();
    int findNextSync(uint8_t *data, size_t len);
    int sendBytes(uint8_t *data, size_t len);
    void setDecoderItems();
    void compute_audioCurrentTime(int bd);
    void printProcessLog(int r, const char* s = "");    // 新加的
    void printDecodeError(int r);
    void showID3Tag(const char *tag, const char *val);
    size_t readAudioHeader(uint32_t bytes);
    int read_WAV_Header(uint8_t *data, size_t len);
    int read_FLAC_Header(uint8_t *data, size_t len);
    int read_ID3_Header(uint8_t *data, size_t len);
    int read_M4A_Header(uint8_t *data, size_t len);
    size_t process_m3u8_ID3_Header(uint8_t *packet);
    bool setSampleRate(uint32_t hz);
    bool setBitsPerSample(int bits);
    bool setChannels(int channels);
    bool setBitrate(int br);
    bool playChunk();
    bool playSample(int16_t sample[2]);
    void computeVUlevel(int16_t sample[2]);
    void computeLimit();
    int32_t Gain(int16_t s[2]);
    void showstreamtitle(const char *ml);
    bool parseContentType(char *ct);
    bool parseHttpResponseHeader();
    bool initializeDecoder();
    esp_err_t I2Sstart(uint8_t i2s_num);
    esp_err_t I2Sstop(uint8_t i2s_num);
    void urlencode(char *buff, uint16_t buffLen, bool spacesOnly = false);
    int16_t *IIR_filterChain0(int16_t iir_in[2], bool clear = false);
    int16_t *IIR_filterChain1(int16_t iir_in[2], bool clear = false);
    int16_t *IIR_filterChain2(int16_t iir_in[2], bool clear = false);
    inline void setDatamode(uint8_t dm) { m_datamode = dm; }
    inline uint8_t getDatamode() { return m_datamode; }
    inline uint32_t streamavail() { return _client ? _client->available() : 0; }
    void IIR_calculateCoefficients(int8_t G1, int8_t G2, int8_t G3);
    bool ts_parsePacket(uint8_t *packet, uint8_t *packetStart, uint8_t *packetLength);

    //+++ W E B S T R E A M  -  H E L P   F U N C T I O N S +++
    uint16_t readMetadata(uint16_t b, bool first = false);
    size_t chunkedDataTransfer(uint8_t *bytes);
    bool readID3V1Tag();
    boolean streamDetection(uint32_t bytesAvail);
    void seek_m4a_stsz();
    void seek_m4a_ilst();
    uint32_t m4a_correctResumeFilePos(uint32_t resumeFilePos);
    uint32_t flac_correctResumeFilePos(uint32_t resumeFilePos);
    uint32_t mp3_correctResumeFilePos(uint32_t resumeFilePos);
    uint8_t determineOggCodec(uint8_t *data, uint16_t len);

    //++++ implement several function with respect to the index of string ++++
    void strlower(char *str)
    {
        unsigned char *p = (unsigned char *)str;
        while (*p)
        {
            *p = tolower((unsigned char)*p);
            p++;
        }
    }

    void trim(char *s)
    {
        // fb   trim in place
        char *pe;
        char *p = s;
        while (isspace(*p))
            p++; // left
        pe = p;  // right
        while (*pe != '\0')
            pe++;
        do
        {
            pe--;
        } while ((pe > p) && isspace(*pe));
        if (p == s)
        {
            *++pe = '\0';
        }
        else
        { // move
            while (p <= pe)
                *s++ = *p++;
            *s = '\0';
        }
    }

    bool startsWith(const char *base, const char *str)
    {
        // fb
        char c;
        while ((c = *str++) != '\0')
            if (c != *base++)
                return false;
        return true;
    }

    bool endsWith(const char *base, const char *str)
    {
        // fb
        int slen = strlen(str) - 1;
        const char *p = base + strlen(base) - 1;
        while (p > base && isspace(*p))
            p--; // rtrim
        p -= slen;
        if (p < base)
            return false;
        return (strncmp(p, str, slen) == 0);
    }

    int indexOf(const char *base, const char *str, int startIndex = 0)
    {
        // fb
        const char *p = base;
        for (; startIndex > 0; startIndex--)
            if (*p++ == '\0')
                return -1;
        char *pos = strstr(p, str);
        if (pos == nullptr)
            return -1;
        return pos - base;
    }

    int indexOf(const char *base, char ch, int startIndex = 0)
    {
        // fb
        const char *p = base;
        for (; startIndex > 0; startIndex--)
            if (*p++ == '\0')
                return -1;
        char *pos = strchr(p, ch);
        if (pos == nullptr)
            return -1;
        return pos - base;
    }

    int lastIndexOf(const char *haystack, const char *needle)
    {
        // fb
        int nlen = strlen(needle);
        if (nlen == 0)
            return -1;
        const char *p = haystack - nlen + strlen(haystack);
        while (p >= haystack)
        {
            int i = 0;
            while (needle[i] == p[i])
                if (++i == nlen)
                    return p - haystack;
            p--;
        }
        return -1;
    }

    int lastIndexOf(const char *haystack, const char needle)
    {
        // fb
        const char *p = strrchr(haystack, needle);
        return (p ? p - haystack : -1);
    }

    int specialIndexOf(uint8_t *base, const char *str, int baselen, bool exact = false)
    {
        int result = 0; // seek for str in buffer or in header up to baselen, not nullterninated
        if (strlen(str) > baselen)
            return -1; // if exact == true seekstr in buffer must have "\0" at the end
        for (int i = 0; i < baselen - strlen(str); i++)
        {
            result = i;
            for (int j = 0; j < strlen(str) + exact; j++)
            {
                if (*(base + i + j) != *(str + j))
                {
                    result = -1;
                    break;
                }
            }
            if (result >= 0)
                break;
        }
        return result;
    }

    // some other functions
    size_t bigEndian(uint8_t *base, uint8_t numBytes, uint8_t shiftLeft = 8)
    {
        uint64_t result = 0;
        if (numBytes < 1 || numBytes > 8)
            return 0;
        for (int i = 0; i < numBytes; i++)
        {
            result += *(base + i) << (numBytes - i - 1) * shiftLeft;
        }
        if (result > SIZE_MAX)
        {
            log_e("range overflow");
            result = 0;
        } // overflow
        return (size_t)result;
    }

    bool b64encode(const char *source, uint16_t sourceLength, char *dest)
    {
        size_t size = base64_encode_expected_len(sourceLength) + 1;
        char *buffer = (char *)malloc(size);
        if (buffer)
        {
            base64_encodestate _state;
            base64_init_encodestate(&_state);
            int len = base64_encode_block(&source[0], sourceLength, &buffer[0], &_state);
            base64_encode_blockend((buffer + len), &_state);
            memcpy(dest, buffer, strlen(buffer));
            dest[strlen(buffer)] = '\0';
            free(buffer);
            return true;
        }
        return false;
    }
    size_t urlencode_expected_len(const char *source)
    {
        size_t expectedLen = strlen(source);
        for (int i = 0; i < strlen(source); i++)
        {
            if (isalnum(source[i]))
            {
                ;
            }
            else
                expectedLen += 2;
        }
        return expectedLen;
    }
    void vector_clear_and_shrink(vector<char *> &vec)
    {
        uint size = vec.size();
        for (int i = 0; i < size; i++)
        {
            if (vec[i])
            {
                free(vec[i]);
                vec[i] = NULL;
            }
        }
        vec.clear();
        vec.shrink_to_fit();
    }
    uint32_t simpleHash(const char *str)
    {
        if (str == NULL)
            return 0;
        uint32_t hash = 0;
        for (int i = 0; i < strlen(str); i++)
        {
            if (str[i] < 32)
                continue; // ignore control sign
            hash += (str[i] - 31) * i * 32;
        }
        return hash;
    }

private:
    const char *codecname[10] = {"unknown", "WAV", "MP3", "AAC", "M4A", "FLAC", "AACP", "OPUS", "OGG", "VORBIS"};
    enum : int
    {
        APLL_AUTO = -1,
        APLL_ENABLE = 1,
        APLL_DISABLE = 0
    };
    enum : int
    {
        EXTERNAL_I2S = 0,
        INTERNAL_DAC = 1,
        INTERNAL_PDM = 2
    };
    enum : int
    {
        FORMAT_NONE = 0,
        FORMAT_M3U = 1,
        FORMAT_PLS = 2,
        FORMAT_ASX = 3,
        FORMAT_M3U8 = 4
    };
    enum : int
    {
        AUDIO_NONE,
        HTTP_RESPONSE_HEADER,
        AUDIO_DATA,
        AUDIO_LOCALFILE,
        AUDIO_PLAYLISTINIT,
        AUDIO_PLAYLISTHEADER,
        AUDIO_PLAYLISTDATA
    };
    enum : int
    {
        FLAC_BEGIN = 0,
        FLAC_MAGIC = 1,
        FLAC_MBH = 2,
        FLAC_SINFO = 3,
        FLAC_PADDING = 4,
        FLAC_APP = 5,
        FLAC_SEEK = 6,
        FLAC_VORBIS = 7,
        FLAC_CUESHEET = 8,
        FLAC_PICTURE = 9,
        FLAC_OKAY = 100
    };
    enum : int
    {
        M4A_BEGIN = 0,
        M4A_FTYP = 1,
        M4A_CHK = 2,
        M4A_MOOV = 3,
        M4A_FREE = 4,
        M4A_TRAK = 5,
        M4A_MDAT = 6,
        M4A_ILST = 7,
        M4A_MP4A = 8,
        M4A_AMRDY = 99,
        M4A_OKAY = 100
    };
    enum : int
    {
        CODEC_NONE = 0,
        CODEC_WAV = 1,
        CODEC_MP3 = 2,
        CODEC_AAC = 3,
        CODEC_M4A = 4,
        CODEC_FLAC = 5,
        CODEC_AACP = 6,
        CODEC_OPUS = 7,
        CODEC_OGG = 8,
        CODEC_VORBIS = 9
    };
    enum : int
    {
        ST_NONE = 0,
        ST_WEBFILE = 1,
        ST_WEBSTREAM = 2
    };
    typedef enum
    {
        LEFTCHANNEL = 0,
        RIGHTCHANNEL = 1
    } SampleIndex;
    typedef enum
    {
        LOWSHELF = 0,
        PEAKEQ = 1,
        HIFGSHELF = 2
    } FilterType;

    typedef struct _filter
    {
        float a0;
        float a1;
        float a2;
        float b1;
        float b2;
    } filter_t;

    typedef struct _pis_array
    {
        int number;
        int pids[4];
    } pid_array;

    File audiofile;                // @suppress("Abstract class cannot be instantiated")
    WiFiClient client;             // @suppress("Abstract class cannot be instantiated")
    WiFiClientSecure clientsecure; // @suppress("Abstract class cannot be instantiated")
    WiFiClient *_client = nullptr;
    SemaphoreHandle_t mutex_audio;
    i2s_config_t m_i2s_config = {}; // stores values for I2S driver
    i2s_pin_config_t m_pin_config = {};
    std::vector<char *> m_playlistContent; // m3u8 playlist buffer
    std::vector<char *> m_playlistURL;     // m3u8 streamURLs buffer
    std::vector<uint32_t> m_hashQueue;

    const size_t m_frameSizeWav = 1024;
    const size_t m_frameSizeMP3 = 1600;
    const size_t m_frameSizeAAC = 1600;
    const size_t m_frameSizeFLAC = 4096 * 4;
    const size_t m_frameSizeOPUS = 1024;
    const size_t m_frameSizeVORBIS = 4096 * 2;

    static const uint8_t m_tsPacketSize = 188;
    static const uint8_t m_tsHeaderSize = 4;

    char *m_ibuff = nullptr; // used in audio_info()
    char *m_chbuf = NULL;
    uint16_t m_chbufSize = 0;               // will set in constructor (depending on PSRAM)
    char *m_lastHost = NULL;                // Store the last URL to a webstream
    char *m_playlistBuff = NULL;            // stores playlistdata
    const uint16_t m_plsBuffEntryLen = 256; // length of each entry in playlistBuff
    filter_t m_filter[3];                   // digital filters
    int m_LFcount = 0;                      // Detection of end of header
    uint32_t m_sampleRate = 16000;
    uint32_t m_bitRate = 0;       // current bitrate given fom decoder
    uint32_t m_avr_bitrate = 0;   // average bitrate, median computed by VBR
    int m_readbytes = 0;          // bytes read
    uint32_t m_metacount = 0;     // counts down bytes between metadata
    int m_controlCounter = 0;     // Status within readID3data() and readWaveHeader()
    int8_t m_balance = 0;         // -16 (mute left) ... +16 (mute right)
    uint16_t m_vol = 100;         // volume改了（21改成100）
    uint8_t m_vol_steps = 100;    // default改了（21改成100）
    double m_limit_left = 0;      // limiter 0 ... 1, left channel
    double m_limit_right = 0;     // limiter 0 ... 1, right channel
    uint8_t m_curve = 0;          // volume characteristic
    uint8_t m_bitsPerSample = 16; // bitsPerSample
    uint8_t m_channels = 2;
    uint8_t m_i2s_num = I2S_NUM_0;          // I2S_NUM_0 or I2S_NUM_1
    uint8_t m_playlistFormat = 0;           // M3U, PLS, ASX
    uint8_t m_codec = CODEC_NONE;           //
    uint8_t m_expectedCodec = CODEC_NONE;   // set in connecttohost (e.g. http://url.mp3 -> CODEC_MP3)
    uint8_t m_expectedPlsFmt = FORMAT_NONE; // set in connecttohost (e.g. streaming01.m3u) -> FORMAT_M3U)
    uint8_t m_filterType[2];                // lowpass, highpass
    uint8_t m_streamType = ST_NONE;
    uint8_t m_ID3Size = 0;     // lengt of ID3frame - ID3header
    uint8_t m_vuLeft = 0;      // average value of samples, left channel
    uint8_t m_vuRight = 0;     // average value of samples, right channel
    int16_t *m_outBuff = NULL; // Interleaved L/R
    int16_t m_validSamples = 0;
    int16_t m_curSample = 0;
    int16_t m_decodeError = 0;      // Stores the return value of the decoder
    uint16_t m_datamode = 0;        // Statemaschine
    uint16_t m_streamTitleHash = 0; // remember streamtitle, ignore multiple occurence in metadata
    uint16_t m_timeout_ms = 250;
    uint16_t m_timeout_ms_ssl = 2700;
    uint8_t m_flacBitsPerSample = 0;         // bps should be 16
    uint8_t m_flacNumChannels = 0;           // can be read out in the FLAC file header
    uint32_t m_flacSampleRate = 0;           // can be read out in the FLAC file header
    uint16_t m_flacMaxFrameSize = 0;         // can be read out in the FLAC file header
    uint16_t m_flacMaxBlockSize = 0;         // can be read out in the FLAC file header
    uint32_t m_flacTotalSamplesInStream = 0; // can be read out in the FLAC file header
    uint32_t m_metaint = 0;                  // Number of databytes between metadata
    uint32_t m_chunkcount = 0;               // Counter for chunked transfer
    uint32_t m_t0 = 0;                       // store millis(), is needed for a small delay
    uint32_t m_contentlength = 0;            // Stores the length if the stream comes from fileserver
    uint32_t m_bytesNotDecoded = 0;          // pictures or something else that comes with the stream
    uint32_t m_PlayingStartTime = 0;         // Stores the milliseconds after the start of the audio
    int32_t m_resumeFilePos = -1;            // the return value from stopSong() can be entered here, (-1) is idle
    int32_t m_fileStartPos = -1;             // may be set in connecttoFS()新加的
    uint16_t m_m3u8_targetDuration = 10;     //
    uint32_t m_stsz_numEntries = 0;          // num of entries inside stsz atom (uint32_t)
    uint32_t m_stsz_position = 0;            // pos of stsz atom within file
    bool m_f_metadata = false;               // assume stream without metadata
    bool m_f_unsync = false;                 // set within ID3 tag but not used
    bool m_f_exthdr = false;                 // ID3 extended header
    bool m_f_ssl = false;
    bool m_f_running = false;
    bool m_f_firstCall = false;         // InitSequence for processWebstream and processLokalFile
    bool m_f_chunked = false;           // Station provides chunked transfer
    bool m_f_firstmetabyte = false;     // True if first metabyte (counter)
    bool m_f_playing = false;           // valid mp3 stream recognized
    bool m_f_tts = false;               // text to speech
    bool m_f_loop = false;              // Set if audio file should loop
    bool m_f_forceMono = false;         // if true stereo -> mono
    bool m_f_internalDAC = false;       // false: output vis I2S, true output via internal DAC
    bool m_f_rtsp = false;              // set if RTSP is used (m3u8 stream)
    bool m_f_m3u8data = false;          // used in processM3U8entries
    bool m_f_Log = false;               // set in platformio.ini  -DAUDIO_LOG and -DCORE_DEBUG_LEVEL=3 or 4
    bool m_f_continue = false;          // next m3u8 chunk is available
    bool m_f_ts = true;                 // transport stream
    bool m_f_m4aID3dataAreRead = false; // has the m4a-ID3data already been read?
    uint8_t m_f_channelEnabled = 3;     // internal DAC, both channels
    uint32_t m_audioFileDuration = 0;
    float m_audioCurrentTime = 0;
    uint32_t m_audioDataStart = 0;  // in bytes
    size_t m_audioDataSize = 0;     //
    float m_filterBuff[3][2][2][2]; // IIR filters memory for Audio DSP
    float m_corr = 1.0;             // correction factor for level adjustment
    size_t m_i2s_bytesWritten = 0;  // set in i2s_write() but not used
    size_t m_file_size = 0;         // size of the file
    size_t m_fileSize = 0;          // size of the file新加的
    uint16_t m_filterFrequency[2];
    int8_t m_gain0 = 0; // cut or boost filters (EQ)
    int8_t m_gain1 = 0;
    int8_t m_gain2 = 0;

    pid_array m_pidsOfPMT;
    int16_t m_pidOfAAC;
    uint8_t m_packetBuff[m_tsPacketSize];
    int16_t m_pesDataLength = 0;
};

//----------------------------------------------------------------------------------------------------------------------
